hi michael, transferred the call using callmanager. the weird thing is, the unusual latency is present only in x-lite and not present in windoze messenger. i played around and changed the codecs but didn't find anything unusual... the root of the problem lies in the call hold, when i place a call on hold, upon resume, the audio becomes lagged
when using messenger, the audio initially is lagged but is able to catch up. a series similar to this line comes up on console upon resume I/O 78216400 160 320 Late 4 4 14.632 0.020 14.715 0.020 0.083 ----- Original Message ----- From: "Michael Manousos" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, October 09, 2003 9:25 PM Subject: Re: [Asterisk-Users] 5 second latency sip to oh323 > > How do you transfer the call? > > Michael. > > > Kelvin Chua wrote: > > hi guys, > > > > i'm using sept 30 cvs and oh323 5.5 > > > > i'm having 5 second latecy(on only 1 audio path) when a call is > > transferred.... > > the scenario is this: > > > > sip--------->asterisk----->h323:operator (who then transfers the call) > > > > ---------------->h323:destination > > > > ------------------audio path 5-second latency----------------> > > <------------------------audio path > > ok------------------------------- > > > > > > > > > > here is the output of the "show channels" > > > > H323:19742 (voip s 1 ) Up Bridged Call > > SIP/kelvin-6952 > > SIP/kelvin-6952 (voip 2010 1 ) Up Dial > > OH323/H323:[EMAIL PROTECTED]|25|mt > > > > > > > > the problem only exists in transferred calls > > any info would be appreciated thanks =) > > > > ~kelvin > > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
